Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.
BUG=1298
Review URL: https://webrtc-codereview.appspot.com/1060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 0438510..5340ed2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -60,11 +60,13 @@
protected:
RtcpFormatRembTest()
: over_use_detector_options_(),
+ system_clock_(Clock::GetRealTimeClock()),
remote_bitrate_observer_(),
remote_bitrate_estimator_(RemoteBitrateEstimator::Create(
- &remote_bitrate_observer_,
over_use_detector_options_,
- RemoteBitrateEstimator::kMultiStreamEstimation)) {}
+ RemoteBitrateEstimator::kMultiStreamEstimation,
+ &remote_bitrate_observer_,
+ system_clock_)) {}
virtual void SetUp();
virtual void TearDown();
@@ -79,7 +81,6 @@
};
void RtcpFormatRembTest::SetUp() {
- system_clock_ = Clock::GetRealTimeClock();
RtpRtcp::Configuration configuration;
configuration.id = 0;
configuration.audio = false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index e300439..d05dd2d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -113,6 +113,19 @@
return _lastReceived;
}
+WebRtc_Word64
+RTCPReceiver::LastReceivedReceiverReport() const {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ WebRtc_Word64 last_received_rr = -1;
+ for (ReceivedInfoMap::const_iterator it = _receivedInfoMap.begin();
+ it != _receivedInfoMap.end(); ++it) {
+ if (it->second->lastTimeReceived > last_received_rr) {
+ last_received_rr = it->second->lastTimeReceived;
+ }
+ }
+ return last_received_rr;
+}
+
WebRtc_Word32
RTCPReceiver::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
{
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 08ff37b..befe2df 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -37,6 +37,7 @@
WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
WebRtc_Word64 LastReceived();
+ WebRtc_Word64 LastReceivedReceiverReport() const;
void SetSSRC( const WebRtc_UWord32 ssrc);
void SetRelaySSRC( const WebRtc_UWord32 ssrc);
@@ -197,6 +198,8 @@
RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
private:
+ typedef std::map<WebRtc_UWord32, RTCPHelp::RTCPReceiveInformation*>
+ ReceivedInfoMap;
WebRtc_Word32 _id;
Clock* _clock;
RTCPMethod _method;
@@ -221,8 +224,7 @@
// Received report blocks.
std::map<WebRtc_UWord32, RTCPHelp::RTCPReportBlockInformation*>
_receivedReportBlockMap;
- std::map<WebRtc_UWord32, RTCPHelp::RTCPReceiveInformation*>
- _receivedInfoMap;
+ ReceivedInfoMap _receivedInfoMap;
std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>
_receivedCnameMap;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 783b13a..11d158a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -172,9 +172,10 @@
remote_bitrate_observer_(),
remote_bitrate_estimator_(
RemoteBitrateEstimator::Create(
- &remote_bitrate_observer_,
over_use_detector_options_,
- RemoteBitrateEstimator::kMultiStreamEstimation)) {
+ RemoteBitrateEstimator::kMultiStreamEstimation,
+ &remote_bitrate_observer_,
+ &system_clock_)) {
test_transport_ = new TestTransport();
RtpRtcp::Configuration configuration;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 7bd2931..e3cadbd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -99,13 +99,14 @@
protected:
RtcpSenderTest()
: over_use_detector_options_(),
+ system_clock_(Clock::GetRealTimeClock()),
remote_bitrate_observer_(),
remote_bitrate_estimator_(
RemoteBitrateEstimator::Create(
- &remote_bitrate_observer_,
over_use_detector_options_,
- RemoteBitrateEstimator::kMultiStreamEstimation)) {
- system_clock_ = Clock::GetRealTimeClock();
+ RemoteBitrateEstimator::kMultiStreamEstimation,
+ &remote_bitrate_observer_,
+ system_clock_)) {
test_transport_ = new TestTransport();
RtpRtcp::Configuration configuration;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
index 066b3c2..e128a10 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -15,7 +15,8 @@
namespace webrtc {
enum { kRtpRtcpMaxIdleTimeProcess = 5,
kRtpRtcpBitrateProcessTimeMs = 10,
- kRtpRtcpPacketTimeoutProcessTimeMs = 100 };
+ kRtpRtcpPacketTimeoutProcessTimeMs = 100,
+ kRtpRtcpRttProcessTimeMs = 1000 };
enum { NACK_PACKETS_MAX_SIZE = 256 }; // in packets
enum { NACK_BYTECOUNT_SIZE = 60}; // size of our NACK history
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 2cc17a3..3cb7994 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -11,7 +11,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include <string.h>
-
#include <cassert>
#include "webrtc/common_types.h"
@@ -39,8 +38,6 @@
namespace webrtc {
-const WebRtc_UWord16 kDefaultRtt = 200;
-
static RtpData* NullObjectRtpData() {
static NullRtpData null_rtp_data;
return &null_rtp_data;
@@ -107,6 +104,7 @@
last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
last_packet_timeout_process_time_(
configuration.clock->TimeInMilliseconds()),
+ last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
packet_overhead_(28), // IPV4 UDP.
critical_section_module_ptrs_(
CriticalSectionWrapper::CreateCriticalSection()),
@@ -258,33 +256,30 @@
ProcessDeadOrAliveTimer();
const bool default_instance(child_modules_.empty() ? false : true);
- if (!default_instance && rtcp_sender_.TimeToSendRTCPReport()) {
- WebRtc_UWord16 max_rtt = 0;
+ if (!default_instance) {
if (rtcp_sender_.Sending()) {
- std::vector<RTCPReportBlock> receive_blocks;
- rtcp_receiver_.StatisticsReceived(&receive_blocks);
- for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
- it != receive_blocks.end(); ++it) {
- WebRtc_UWord16 rtt = 0;
- rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
- max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
+ // Process RTT if we have received a receiver report and we haven't
+ // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
+ if (rtcp_receiver_.LastReceivedReceiverReport() >
+ last_rtt_process_time_ && now >= last_rtt_process_time_ +
+ kRtpRtcpRttProcessTimeMs) {
+ last_rtt_process_time_ = now;
+ std::vector<RTCPReportBlock> receive_blocks;
+ rtcp_receiver_.StatisticsReceived(&receive_blocks);
+ uint16_t max_rtt = 0;
+ for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
+ it != receive_blocks.end(); ++it) {
+ uint16_t rtt = 0;
+ rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
+ max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
+ }
+ // Report the rtt.
+ if (rtt_observer_ && max_rtt != 0)
+ rtt_observer_->OnRttUpdate(max_rtt);
}
- // Report the rtt.
- if (rtt_observer_ && max_rtt != 0)
- rtt_observer_->OnRttUpdate(max_rtt);
- } else {
- // No valid RTT estimate, probably since this is a receive only channel.
- // Use an estimate set by a send module.
- max_rtt = rtcp_receiver_.RTT();
- }
- if (max_rtt == 0) {
- // No own rtt calculation or set rtt, use default value.
- max_rtt = kDefaultRtt;
- }
- // Verify receiver reports are delivered and the reported sequence number is
- // increasing.
- if (rtcp_sender_.Sending()) {
+ // Verify receiver reports are delivered and the reported sequence number
+ // is increasing.
int64_t rtcp_interval = RtcpReportInterval();
if (rtcp_receiver_.RtcpRrTimeout(rtcp_interval)) {
LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
@@ -292,13 +287,8 @@
LOG_F(LS_WARNING) <<
"Timeout: No increase in RTCP RR extended highest sequence number.";
}
- }
- if (remote_bitrate_) {
- // TODO(mflodman) Remove this and let this be propagated by CallStats.
- remote_bitrate_->SetRtt(max_rtt);
- remote_bitrate_->UpdateEstimate(rtp_receiver_->SSRC(), now);
- if (TMMBR()) {
+ if (remote_bitrate_ && TMMBR()) {
unsigned int target_bitrate = 0;
std::vector<unsigned int> ssrcs;
if (remote_bitrate_->LatestEstimate(&ssrcs, &target_bitrate)) {
@@ -309,7 +299,8 @@
}
}
}
- rtcp_sender_.SendRTCP(kRtcpReport);
+ if (rtcp_sender_.TimeToSendRTCPReport())
+ rtcp_sender_.SendRTCP(kRtcpReport);
}
if (UpdateRTCPReceiveInformationTimers()) {
@@ -1995,22 +1986,6 @@
*nack_rate = rtp_sender_.NackOverheadRate();
}
-int ModuleRtpRtcpImpl::EstimatedReceiveBandwidth(
- WebRtc_UWord32* available_bandwidth) const {
- if (remote_bitrate_) {
- std::vector<unsigned int> ssrcs;
- if (!remote_bitrate_->LatestEstimate(&ssrcs, available_bandwidth)) {
- return -1;
- }
- if (!ssrcs.empty()) {
- *available_bandwidth /= ssrcs.size();
- }
- return 0;
- }
- // No bandwidth receive-side bandwidth estimation is connected to this module.
- return -1;
-}
-
// Bad state of RTP receiver request a keyframe.
void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
RequestKeyFrame();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 5883a11..666612b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -427,9 +427,6 @@
WebRtc_UWord32* fec_rate,
WebRtc_UWord32* nackRate) const;
- virtual int EstimatedReceiveBandwidth(
- WebRtc_UWord32* available_bandwidth) const;
-
virtual void SetRemoteSSRC(const WebRtc_UWord32 ssrc);
virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 send_report);
@@ -493,6 +490,7 @@
WebRtc_Word64 last_process_time_;
WebRtc_Word64 last_bitrate_process_time_;
WebRtc_Word64 last_packet_timeout_process_time_;
+ WebRtc_Word64 last_rtt_process_time_;
WebRtc_UWord16 packet_overhead_;
scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
